HI ,
Is there Any way is there so that I can store my recordings directly to
a database rather storing the same to a file .
Thanks in advance .
Regards
Mahesh
--
_
-- Bandwidth and Colocation Provided by
Hi,
Excuse me if I'm late to reply but my first response has been blocked by the
moderator (message too big)
So I've created an account on rapidshare to share my config.log and
menuselect/config.log
Hope it will help.
The link : http://rapidshare.com/users/Z8SX25
Thanks for any help !
Danny, thank you!
On Wed, Sep 22, 2010 at 10:31 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham
*Sent:* Wednesday, September 22, 2010 4:28 PM
*To:* Asterisk Users
Hello,
I want to graphically display the number of calls per minute to an
extension.
The programs I have found it possible to do so but the average is done on
time or day ...
I use Mysql CDR
Thank you,
Mickael
--
_
--
use CACTI
On 9/23/2010 3:10 PM, Mickael MONSIEUR wrote:
Hello,
I want to graphically display the number of calls per minute to an
extension.
The programs I have found it possible to do so but the average is done
on time or day ...
I use Mysql CDR
Thank you,
Mickael
--
http://forums.cacti.net/viewtopic.php?p=111317
Thank you.
2010/9/23 Faisal Hanif fai...@vopium.com
use CACTI
On 9/23/2010 3:10 PM, Mickael MONSIEUR wrote:
Hello,
I want to graphically display the number of calls per minute to an
extension.
The programs I have found it possible to do
Hi All;
I have my friend that use his mobile (Nimbuz) to connect for the Asterisk and
his account was working fine. Suddenly it stop working (not able to register).
From my mobile (Nokia) I was able to register using my username and password,
so I tried to register using his (my friend)
Greetings,
Because of the heavy load and the high expectations of an asterisk server
offered as a solution integrated with our CRM software.. we were looking
into other possibilities than software Licenses for G729 and G723 codecs..
to lower the pressure on the processor giving it more space to do
Hi!
And the third hit in my google result is this:
http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
Since I mentioned in my previous message that you will find the answer
in the archive of this list you could have found that even without
google. gmane.org for
I hope it cant be done using dialplan but it can be done through AGI
scripting...you can use your favorite programming language PHP / Perl and
try to record the call with record application and point to the database
instead of a folder path.
Asterisk Users your feedback also welcome on this..
On
23.09.2010 16:06, Philipp von Klitzing пишет:
Hi!
And the third hit in my google result is this:
http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
Since I mentioned in my previous message that you will find the answer
in the archive of this list you could have
Hi!
There are 2 things I can't understand
- 1. how can I know channel name?
${CHANNEL}
2. where should I call this SHARED function? before Dial, after Dial?
Either In the macro that you specify using the M option of Dial() or in
the h extension. You will, however, have trouble treating the
Thanks ,
Do you have some sample for that .
Regards
Mahesh
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext
Gopalakrishnan A.N
Sent: Thursday, September 23, 2010 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial
sorry I dont have samples, but you can find lots of php agi scripts by
googling..
On Thu, Sep 23, 2010 at 7:07 PM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
Thanks ,
Do you have some sample for that .
Regards
Mahesh
*From:*
Just my .02 -
#1 - voip-info.org is a good resource for finding this kind of script
#2 - I think you are asking for trouble doing a direct recording using the
PHP/MySQL combination; Personally I would do C/MySQL but the exposure you
face depends on the length of your recording(s).
#3 - since
Dear, I have this scenario:
- PBX Asterisk 1.6.2.10 with private IP 192.168.0.10
- Behind a Cisco ASA firewall that connects to Internet
- SIP trunk to Net2Phone with these parameters (nat=no):
host=200.58.113.60
username=DOLLY
secret=123456
port=5060
type=peer
dtmfmode=rfc2833
disallow=all
On Wed, Sep 22, 2010 at 10:46 AM, Adam Moffett a...@plexicomm.net wrote:
That's probably what I'm going to have to do. Thanks.
I suppose that merely removing ATA and asterisk from the middle, and
plugging a pots line into a fax machine is out of the question.
If you are running 1.6,
- Tarek Sawah tareksa...@hotmail.com wrote:
Greetings,
Because of the heavy load and the high expectations of an asterisk
server
offered as a solution integrated with our CRM software.. we were
looking
into other possibilities than software Licenses for G729 and G723
codecs..
to lower
Hi Guys,
i could turn debug on in a asterisk 1.6 box (by enabling debug in
logger.conf and core set debug to 0), but my issue is i cannot enable
debugging in a 1.2 box by doing the same 2 steps, also this is a production
server so i can't restart with debug enabled, do you guys know how i can
Please give a try like the following,
Xlite (Configure Net2Phone A/c) --- Cisco ASA Firewall --- Internet cloud,
if the above works then there is no problem with your firewall, replace the
nat=yes and canreinvite=yes
otherwise you have to allow the ports 5060 (TCP), 5000 to 3 (UDP) in
your
Thanks .
I checked in voip info but I was not able to find a script which records
directly to database . All scripts were recording directly to files .
About #2 and #3 can you please provide some more information .
Regards
Mahesh
From: asterisk-users-boun...@lists.digium.com
Is there a way to specify which IP address to originate calls from in a peer
on sip.conf?
I need to send calls from 10.1.3.10 which is a routed network through
openvpn, but it's using 10.39.0.10 which is a vpn IP address - the asterisk
box is the same box as the vpn bridge for the 10.1.3.0/24
From what you explained it seems to me that your mobile provider has blocked
your sip communication altogether. Have you tried changing IP address of
your asterisk server? If changing IP works, then probably your provider has
blocked you sip communication by IP only.
Zeeshan A Zakaria
--
We can resell the Sangoma card. They have some higher license counts as
well.
They are also offering a step up offering. If you buy at one level and need
to move to the next.
They will offer you a trade back on the old card.
Bryant
From: Tim Nelson
On 09/23/2010 06:48 AM, Tarek Sawah wrote:
Greetings,
Because of the heavy load and the high expectations of an asterisk server
offered as a solution integrated with our CRM software.. we were looking
into other possibilities than software Licenses for G729 and G723 codecs..
to lower the
Hi,
Using 1.6.2.13.
I'd like to know how I can force Asterisk to fork a call. To simplify
things, Let's say I have an out context (for outbound calls) and an in (for
inbound). If person A wants to call person B, and both are on my servers, I
don`t want to send the call out. I want all
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext Danny
Nicholas
Sent: Thursday, September 23, 2010 7:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Record() Cmd and My SQL
On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
HI ,
Is there Any way is there so that I can store my recordings directly to a
database rather storing the same to a file .
Please, please, please tell us why you would want to do that.
--
I'm coming to Asterisk from a traditional PSTN environment, so forgive
me if I use the wrong Asterisk/SIP terminology.
I need to make a product where calls come in go through various menus
and based on various configurations perform attended transfers, blind
transfers, and patch callers
Hi
Isn't there any way to configure the username in the hardphone to be
just ?
Yes.there is no way to cofigure as in the hardphone.It will cost and
spend time a lot to implement
thanks.
Date: Tue, 21 Sep 2010 14:04:17 +0100
From: s...@open-t.co.uk
To:
On 23 Sep 2010, at 17:29, t. k wrote:
Isn't there any way to configure the username in the hardphone to be
just ?
Yes.there is no way to cofigure as in the hardphone.It will cost and
spend time a lot to implement
Then I think the short answer is that it's not compatible.
Steve
--
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of Danny Nicholas
Sent: Wednesday, September 22, 2010 5:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk- speech to text(Voicemail
On 09/23/2010 06:48 AM, Tarek Sawah wrote:
Greetings,
Because of the heavy load and the high expectations of an asterisk
server
offered as a solution integrated with our CRM software.. we were looking
into other possibilities than software Licenses for G729 and G723
codecs..
to lower the
I don't think it's an endpoint issue. I think the SIP packet headers get
over-written by the tunnel (openvpn) protocol.
I'd be rather astonished if OpenVPN itself were responsible for this.
As far as I know, OpenVPN doesn't do higher-level-protocol rewriting
of any sort. It just provides the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin
Sherrill
Sent: Thursday, September 23, 2010 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk-
Hello everyone,
I'm using redfone fonebridge to have Pstn
connectivity on my asterisk box.
I can receive in coming calls however outgoing calls don't go to
provider. It's seems it's a span config problem. Because in systemconf
when I try to config span as follow
The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.
* Make AMI
No, I do not think that my provider blocked my IP address, because I am able to
register for the Asterisk (at that IP address) from an IP Phone, but not from
the mobile. It is well known that the mobile use the digest authentication
(realm) which is not used in the IP Phone.
Any advise?
Bilal,
If you are using 3G or Wifi with your Nokia Native SIP Client.. try to
connect via an internet connection sharing machine.. it seems that your ISP
is blocking INBOUND SIP packets.
Test and let me know
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
softphone, all rtp traffic ceases! Now, a version which does work is
r281875, this
Hi,
I have a server with multiple IP address, Asterisk binding with all of them.
I'd like Asterisk to reply to a SIP peer from the same IP address as the
peer used to register to Asterisk (as opposed to using the main IP address
all the time regardless of how the peer communicated with
Thanks for the detailed info. Problem was solved by including Server B
subnet as the localnet of the Server A (OpenVPN server) and setting each
extension NAT=NO.
Your points are good guides for future problem diagnoses.
Thanks again,
Bruce
On Thu, Sep 23, 2010 at 1:56 PM, Dave Platt
The reason is when doing a load balancing , We cannot confine the
recording to a particular asterisk machine ( If we have more than one
asterisk machine in the topology ).
So a centralized mechanism might be better . So that any machine can
access the recording .
Regards
Mahesh
-Original
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